In mobile communications networks there often occur situations in which the route of a communications connection must be changed e.g. because the location of the mobile station changes. The communications connections of a mobile station, most often calls, are usually routed via the nearest base station. When the mobile station moves and the distance to the serving base station grows, resulting e.g. in the weakening of the radio signal, the mobile network carries out a handover, i.e. the communications connections of the mobile station are switched over to a new, usually the nearest, base station.
For a handover to be successful the target base station needs to have free channels both over the air interface between the mobile station and base station and from the base station to the network. In addition, the data transfer rates of these channels must be suitable. Either the transfer rates must be the same as in the original base station or the mobile station and target base station need to negotiate new speech coding and other coding methods to be subsequently used in the handling of this particular connection in the mobile network. For example, if the target base station is able to set up a connection on a channel the channel rate of which is lower than the original, the mobile station and mobile network need to find a common speech coding method compatible with the lower transfer capacity. Mobile stations, for instance, may use two different speech coding methods which produce data streams that require different transfer rates. The task of the speech coding method is to pack the digitized speech that requires a 64-kbps transfer rate into a format that requires a transfer rate not higher than 13 kbps in the GSM network (Global System for Mobile communications).
In mobile networks according to the prior art there are two kinds of channel over the air interface: half rate (HR) channels and full rate (FR) channels. These terms always refer to the channel rate of the air interface. A base station may support either half rate or full rate channels or both. The capacity of a channel over the air interface is used, apart from the transfer of encoded information, such as e.g. speech, also for channel coding. The purpose of channel coding is to improve the quality of the data transferred over the air interface. For example, certain errors in the transfer can be corrected by means of channel coding and the altered data need not be re-transmitted. However, channel coding adds to the data that must be transferred and, for example, the more there is noise, the more heavier the channel coding that must be used and the greater the part of the transfer capacity that is used for the transfer of channel coding information. Speech coding methods are often classified as compatible with either half rate or full rate channels, depending on the transfer rate required by the transfer over the air interface of encoded speech produced by them.
FIG. 1 illustrates a handover according to the prior art in a GSM network. A mobile station (MS) 101 is connected over the air interface to a base station (BTS1) 102. Channel coding is a function between the mobile station and base station, and it is denoted by an arrow in the lower part of FIG. 1. The base station is connected through a fixed line to a base station controller (BSC) 103, which also controls a second base station (BTS2) 104. The base station controller is connected through a network-side transcoder and rate adaptation unit (TRAU) 105 to a mobile switching center (MSC) 106. GSM speech coding is a function between the terminal and the network-side TRAU, and it is denoted by an arrow in the lower part of FIG. 1. Encoded speech is transferred over a connection from the base station via a base station controller to the TRAU in TRAU frames at a speed which usually is 8 or 16 kbps. This transfer between the base station and TRAU is called base station transmission. Between the TRAU and mobile switching center, speech travels in the same format that is used in fixed telephone networks, and the transfer rate is 64 kbps.
In the situation depicted in FIG. 1 the call or other communications connection uses channel CH1 in base station 102. The mobile network decides to carry out a handover, and intends to move the communications connection over to channel CH2 in base station 104. The switching function 107 in the base station controller shown in FIG. 1 is responsible for carrying out the handover of the connection to the new base station. FIG. 1 illustrates an intra-BSC, inter-cell handover. Intra-cell handovers are also possible, as well as inter-BSC or inter-MSC handovers. In this context we will concentrate on intra-BSC handovers which take place not only when the location of a mobile station changes but also when a half rate channel is to be changed into a full rate channel if, for example, the quality of the radio path is so poor that satisfactory audio quality cannot be guaranteed using channel coding and speech coding methods compatible with a half rate channel. In an intra-cell handover the channel speed may also be changed from full rate to half rate if allowed by the radio path quality and the base station cell lacks free channels.
In mobile networks according to the prior art, a change in the channel rate of a channel over the air interface will result in a change in the speech coding method and possibly in the channel rate of base station transmission, when conventional speech coding methods are used. This is due to the fact that half rate speech coding methods do not support a full rate air interface, and full rate speech coding methods do not support a half rate air interface. In GSM networks, full rate and enhanced full rate (EFR) speech coding methods are usually accompanied by 16-kbps base station transmission, and half rate speech coding methods are usually accompanied by 8-kbps base station transmission. It is also possible to use e.g. 16-kbps base station transmission with half rate speech coding methods, but then some of the transmission capacity will be wasted.
In conventional speech coding methods which produce speech that is coded at a constant rate the speech quality strongly depends on the radio path quality of the connection. When using conventional speech coding methods, the ratio of speech coding and channel coding in the transfer rate over the air interface stays constant all the time. Thus, in noisy conditions the channel coding is not necessarily able to eliminate all spurious effects, which causes uncontrollable degradation in speech quality.
To optimize the speech quality and the transfer capacity in use in a mobile network it is possible to use an adaptive multirate (AMR) system according to the prior art. The AMR system uses a variable-rate speech coding method which is hereinafter called the AMR speech coding method. The term AMR system refers to the whole AMR concept which comprises the measurement of the radio path quality and the selection of a suitable AMR speech coding method and channel coding method.
As the AMR system uses a variable-rate speech coding method, it is possible to vary the ratio of the transfer rates required by the channel coding and speech coding within a framework of a certain air interface channel rate. In the AMR system it is possible in noisy conditions to use a lower speech coding rate and increase the ratio of channel coding in the transfer capacity of a channel over the air interface. When the speech coding method is changed into one that produces encoded speech at a lower rate, the speech quality suffers to a certain extent. However, the speech coding methods are usually designed in such a manner that they produce the best possible speech quality for the transfer rate available. So, by choosing a slower speech coding method and channel coding that better eliminates the degradation of speech quality caused by the air interface it is possible to reduce the speech quality in a controlled manner.
The AMR system according to the prior art also involves air interface channel rate change, not only the optimization of speech and channel coding within the framework of a certain channel rate. Such channel rate change is carried out using the handover described above. The air interface channel rate is changed by means of an intra-cell handover, for example.
FIG. 2 illustrates the operating principle of the AMR system. Like elements in the figures are denoted by like reference designators. The AMR system implementation involves a mobile station 101, base station 102, base station controller 103 and a network-side speech coding unit 105 (i.e. TRAU). In FIG. 2 the AMR speech coding method is represented as a speech encoder 201 in the mobile station and speech coding unit, and as a speech decoder 202 in the same network elements.
In FIG. 2, the horizontal lines coming into and going out of a given block are the block's input and output data streams processed by the block. They do not, however, have any effect on the block's mode of operation. Lines arriving at blocks from above or below mean that the information conveyed through a particular connection does have an effect on the block's operation. A connection carrying speech information is denoted by a thick continuous line in FIG. 2. A downlink connection is shown in the upper part of FIG. 2 and an uplink connection in the lower part of FIG. 2. Intra-channel signalling associated with a downlink connection is denoted by a thin continuous line and signalling associated with an uplink connection is denoted by a thin broken line in FIG. 2.
In addition to blocks 201 and 202 associated with speech coding the mobile station 101 comprises a channel encoder 203 and channel decoder 204 associated with channel coding, and a downlink (DL) connection quality measurement block 205. The base station comprises a channel encoder 203 and decoder 204 as well as an uplink (UL) connection quality measurement block 206. In addition to these, the base station includes a downlink speech codec selection control block 207 and an uplink speech codec selection control block 208. The downlink quality measurement results are sent from the mobile station's block 205 to control block 207, and the uplink quality measurement results are sent from the measurement block 206 to the control block 208 within the base station.
The base station controller comprises two switching fields 209 and 210 through which the connections travel from the base station to the speech coding unit and in the reverse direction. These switching fields belong to the switching function 107. The speech coding unit comprises a speech encoder 201 and decoder 202 associated with speech coding. Selection of downlink speech and channel coding methods is realized as follows. Downlink connection quality is measured in block 205 and the results are delivered to control block 207. The control block informs the speech encoder 201 in the speech coding unit 105 about the speech coding method selected, and the speech encoder will use the selected method to encode the downlink connection. The channel encoder 203 in the base station is informed about the selected speech codec and selects channel coding accordingly. The channel decoder 204 in the mobile station receives signalling information about the channel coding used and can on the basis of this information carry out channel decoding. Also the speech decoder 202 in the mobile station receives signalling information about the speech coding method used.
Speech and channel coding methods for an uplink connection are selected in a similar fashion. The selected speech coding method is signalled via a downlink connection from the control block 208 to the mobile station. The speech encoder 201 and channel encoder 203 in the mobile station receive the information and will use the selected coding methods. Information about the speech coding method used is also sent to the channel decoder 204 in the base station and to the speech decoder 202 in the speech coding unit. In the AMR system, the speech coding method and channel coding method can be changed at 40-ms intervals, i.e. at intervals of two 20-ms speech frames.
At its simplest the switching function 107 in the base station controller in accordance with the prior art is a change-over switch between the original and target channels. In that case, a two-way data stream is switched at once from the original channel to the target channel. The switching moment must be chosen such that the break in the communications connection is as short as possible. Usually the change-over is carried out when the mobile station has been successfully handed over to the target channel in the target base station. Information about a successful change comes in a signalling message. Switching function has to be used always when the air interface channel rate and, consequently, the speech processing method are changed in conjunction with a handover. The network-side speech coding unit 105 changes speech coding methods synchronized with the mobile station at the switching moment.
In certain situations the downlink data stream to the terminal may be branched. In that case, the downlink data stream may during a handover travel to both the original channel and the target channel simultaneously. A corresponding operation on an uplink data stream coming to the base station would be summing, but two encoded audio streams cannot be summed. The switching function must always be used in a handover for an uplink data stream when the data stream is comprised of encoded speech.
FIG. 3 illustrates the operation of the branching/switching function in a base station controller in accordance with the prior art. The uplink and downlink connections are shown as separate one-way connections in the figure. The figure shows four different phases of handover, and like elements are denoted by like reference designators in these subfigures. FIG. 3a illustrates the situation in the beginning of a handover. The base station controller's switching/branching block 300 receives from the speech coding unit a downlink data stream 307 which travels through the block 300 as such to the original base station BTS1 through a downlink connection 301. From the original base station on channel CH1 an uplink data stream 302 travels via switch 305 to the speech coding unit of the mobile network through connection 307 and from there via a mobile switching center 106 to the other end of the connection. In the initial state of the handover no data is carried onto the channel CH2 of the target base station BTS2 through the downlink connection 303 or uplink connection 304.
The base station controller decides to perform a handover and when the target base station channel CH2 has been activated the switching/branching block enters the first intermediate state according to FIG. 3b. The downlink data stream 306 coming from the speech coding unit is branched at point 308 so as to travel both to the original base station through connection 301 and to the target base station through connection 303. The uplink connection still travels via the original base station. The second intermediate state shown in FIG. 3c follows when the mobile station has tuned to channel CH2 of the target base station, in which state the uplink data stream is switched to the target base station by means of switch 305. The downlink data stream is still directed to both base stations. When a signalling message indicating a successful handover has been received, the block enters the final state shown in FIG. 3d, in which state the branching of the downlink data stream at point 308 has been removed. The data streams between the mobile station and the other terminal of the connection travel only via the target base station.
A problem with handovers according to the prior art is that the use of the switching function in a downlink connection impairs the quality of the connection, say the speech quality, as it introduces a break in the downlink connection at the switching moment. The length of the break depends on the transmission delay between the switching function at the base station controller and the mobile station. Furthermore, if synchronization of the mobile station to the new channel is delayed e.g. because of interference in the radio interface or if the handover fails and the mobile station has to in to the original channel, the break may be annoyingly long.
When using the branching/switching function, the quality of the downlink connection stays better. With conventional speech coding methods, branching may be used in handovers in which the speech coding method is not changed. In that case, the data stream traveling through both the original and the target base station is processed in the speech coding unit using the same speech codec. When using conventional speech codecs, the speech coding method need not be changed in conjunction with a handover, if the air interface channel rate is not changed in the handover.
With conventional speech coding methods the switching function is needed at least in handovers in which the air interface channel rate changes between half rate and full rate. Handovers, in which the air interface channel rate changes, occur in situations where e.g. the target base station does not support the channel rate of the call or will not let the call have the channel rate in question, in order to maximize the number of calls, for example.
The problem, when using the AMR system, is that air interface interference is compensated for by changing, for example, a half rate air interface channel into a full rate channel. This results in an increase in the number of handovers in which the air interface channel rate changes. Since the speech coding methods in the AMR system, too, are categorized as compatible with half and full rate air interface channel rates, the speech coding method possibly has to be changed in conjunction with a handover into a method not compatible with the original one. Thus the switching function must be used, which degrades the speech quality.
For some of the AMR speech codecs the base station transmission channel rate of 8 kbps is not enough whereas all the conventional half rate speech codecs are compatible with this rate. If, for example, a call is begun using a half rate speech codec, which requires a base station transmission rate of 16 kbps, it is possible at some point in the call to end up in a situation in which the speech quality would be guaranteed by a combination of channel coding and speech coding for which an 8 kbps base station transmission would be enough. The problem is that base station capacity is at that moment unnecessarily allocated to the connection in question and that changing the transmission capacity of a circuit-switched connection often results in a break in the connection.
So, the use of the AMR system may lead to a situation in which the number of breaks in the connection grows larger. It has been suggested that the number of handovers carried out at the AMR system's initiative be limited to about two in a minute per connection, lest the connection be broken too often. However, limiting the number of handovers requires additional logic on the network side and results in that the capabilities of the AMR system as regards optimization of connection quality cannot be fully exploited.